search for: ecasound

hi, I am recording a continuous 24/7 broadcast using ecasound, oggenc and cronolog. because the broadcast is continuous and to not lose any data, I split up the recorded stream into a file for every 30 minutes using cronolog (http://www.cronolog.org). o the chain looks like the following: ecasound | oggenc | cronolog the problem is the following, because...

I posed a question about this a while back, and Kai found the problem and sent this workaround: > ---------- Forwarded message ---------- > From: Kai Vehmanen <kvehmanen at eca.cx> > Date: Thu, Mar 19, 2009 at 3:03 PM > Subject: broken lame/mp3 output (was: Re: Problem with ecasound on CentOS 5.2, x86_64) > To: MHR <mhullrich at gmail.com> > Cc: ecasound-list at lists.sourceforge.net > > Hi, > > : > > Fortunately, a workaround is easy, just add: > > --cut-- > ext-cmd-mp3-output = lame -b %B -s %S -r --little-endian -S - %f > --cut-- &g...

I'm running ices-kh to stream from jack at 64kbps, and also using oggenc (with ecasound via jack) to record the audio to disk at the same time. This is also running at 64kbps. ices is using virtually no cpu (0.0%), but oggenc is using 15-16%. I can't see why there should be such a difference - both are recording the same audio stream in real time at the same bitrate. oggenc is ge...

I'm running ices-kh to stream from jack at 64kbps, and also using oggenc (with ecasound via jack) to record the audio to disk at the same time. This is also running at 64kbps. ices is using virtually no cpu (0.0%), but oggenc is using 15-16%. I can't see why there should be such a difference - both are recording the same audio stream in real time at the same bitrate. oggenc is ge...

...'m guessing this happens when there is an xrun. I've tried increasing the client timeout to 1000 msec, but this doesn't help. I can't increase the buffer size any more - it's at 2048 and increasing it to 4096 stops ices from starting at all. The setup I'm working with is an ecasound session relaying audio between its input and output ports, with its output connected to all the ices clients for the different streams we're running. Then I'm switching the ecasound input between different stream sources (line in, recorded audio, http relay etc.) I want to be able to do thi...

Hi, I searched the Internet to find an answers, but I didn't find anything useful, so I'm turning to you, maybe someone has the answer. I want to make a realtime broadcast from a Linux box. The source is the soundcard's line-in, and it sends the stream to an Icecast server. I would like to have realtime compressor/limiter functionalities on this Linux box, so the outgoing signal

...you to place a large number of automated calls (using Asterisk) using different call scenario files. The key here is consistency. When Asterisk places the outbound call (and answers the inbound call) it will generate a set of tones while recording the return audio path. Once the run has finished Ecasound will run with various filters and noise gates to detect certain amounts of distortion, signal loss, etc. Calls either pass or fail based on how much variation there is in the audio once it has been returned. Of course you can pass audio through any combination of networks - including the PSTN. A...

On Mon, 18 Oct 2004 16:15:44 +0100, EvilOverlord wrote: >> The setup I'm working with is an ecasound session relaying audio between >> its input and output ports, with its output connected to all the ices >> clients for the different streams we're running. Then I'm >> switching the ecasound input between different stream sources (line in, >> recorded audio, http rel...

...xrun. I've tried increasing the client timeout to >> 1000 msec, but this doesn't help. I can't increase the buffer size any >> more - it's at 2048 and increasing it to 4096 stops ices from starting at >> all. >> >> The setup I'm working with is an ecasound session relaying audio between >> its input and output ports, with its output connected to all the ices >> clients for the different streams we're running. Then I'm >> switching the ecasound input between different stream sources (line in, >> recorded audio, http rel...

On Aug 23, 2004, at 9:35 PM, Ralph Giles wrote: > On Mon, Aug 23, 2004 at 11:49:27AM -0700, Eric Smith wrote: > >> I've tracked the problem down to abuse of the C type size_t in >> aplay.c. So I'm pretty sure it's not a problem in oggenc. > > Hmm. Wouldn't have expected that from alsa, though if it's saving > as 'wav' you have to sort of

Please excuse me, I haven't yet dug into the code too far. What I'd like to do: I'm using ecasound to record for 24 hour periods, which pipes the output directly to flac. I would like to be able to decode sections of this (using until/skip) during recording/encoding. I understand currently that flac can't do this, unless I pipe into dd to skip after decoding (which would be an incredible w...

Sounds like you need to fix the problem at the source first: balancing the levels going *into* your mixing board so that they're not all over the place coming out. You'll never fix that with any hardware or software device. I use a program called ecasound to do software dynamics processing, but you'd have to hack liveice pretty extensively to use it in that situation. Software dynamics processing is pretty complex (& processor-intensive). The mini-disc recorder you speak of almost certainly uses a dedicated DSP chip (or even analog AGC). -...

hello, I've just got ices-kh60 running with jack. It's working OK apart from a problem where the latency on the stream creeps up from about 15-20 secs to around 7 minutes. The stream plays OK, but the audio coming out is 7 minutes behind what goes in. I'm not sure how this can be happening, seeing as I don't think there are any buffers that long in the system, but it is.

...otential sources of quality problems, ranging from echo to voice dropouts and everything in between. With VoIP systems the potential for quality problems increases dramatically, often times making it very difficult to identify the source of the problem. Recqual is a utility that uses Asterisk and Ecasound to make comparisons of complete audio paths - whether they be IP, TDM, or any combination. In this Friday's VoIP User's Conference, Kristian Kielhofner will give an introduction to Recqual and provide several examples of its usage in real world contexts. More on Recqual here on Kristian...

On Thursday 14 October 2004 03:11, Andy Baxter wrote: > I'm running ices-kh to stream from jack at 64kbps, and also using oggenc > (with ecasound via jack) to record the audio to disk at the same time. > This is also running at 64kbps. > > ices is using virtually no cpu (0.0%), but oggenc is using 15-16%. I can't > see why there should be such a difference - both are recording the same > audio stream in real time at the sa...

...9;t help. I can't increase the buffer size any > more - it's at 2048 and increasing it to 4096 stops ices from starting at > all. xrun is just when the input thread cannot get scheduled quickly enough, this may be resolved enabling realtime. > The setup I'm working with is an ecasound session relaying audio between > its input and output ports, with its output connected to all the ices > clients for the different streams we're running. Then I'm > switching the ecasound input between different stream sources (line in, > recorded audio, http relay etc.) I want...

...ens when there is an xrun. I've tried increasing the client timeout to > 1000 msec, but this doesn't help. I can't increase the buffer size any > more - it's at 2048 and increasing it to 4096 stops ices from starting at > all. > > The setup I'm working with is an ecasound session relaying audio between > its input and output ports, with its output connected to all the ices > clients for the different streams we're running. Then I'm > switching the ecasound input between different stream sources (line in, > recorded audio, http relay etc.) I want...

...ens when there is an xrun. I've tried increasing the client timeout to > 1000 msec, but this doesn't help. I can't increase the buffer size any > more - it's at 2048 and increasing it to 4096 stops ices from starting at > all. > > The setup I'm working with is an ecasound session relaying audio between > its input and output ports, with its output connected to all the ices > clients for the different streams we're running. Then I'm > switching the ecasound input between different stream sources (line in, > recorded audio, http relay etc.) I want...

On Mon, 2004-11-01 at 03:23, andy wrote: > I just had a look at it and it's still dropping ports - I set a test > running which has ices-kh running from an ecasound process which is > connected to a new source every 10 minutes, and after about 3 hours it had > lost the jack port connection to ecasound. Can you send me the full log? > 2004-11-01 01:28:09] DBUG stream/_output_oggpacket seen new stream, better get headers > [2004-11-01 01:28:09] D...

Hi, It's going to depend a bit on what format you're going to use. If you plan to stream in ogg vorbis, you can use Ices 2.x and accept PCM via standard input. This means you could run a sox effect or ecasound to get and compress the sound from the soundcard before sending it on to ices. Ecasound can make use of LADSPA plugins which opens you to a range of compressors, such as the SWH compressor plugins which I'm told are pretty good. If you want to do MP3 then I don't have any good ideas so...